hello i want to make a dynamic filter with asterisk follow this features - the calls come from customer pass through asterisk filter to check if it's an spam call before to send to gateway gsm if the call is a spam, then the call no go to the gateway but if the call is a real, then the call pass normaly
Hi I need someone who is diligent with Voip Asterisk. We have a Voip freepbx setup on 1.8 For some reason Iptables is blocking our Phone lines. I disable iptables service. All is working 30 min later it automaticaly reenabled iptables no more Phone. So as the freepbx is old. I want to reinstall a new server. We have 2 tenants 5 trunks and 11 phones
...person to do some kamailio development for us. We would like to have kamailio look up the registrar domain and forward all registrations and invites to and from multiple asterisk servers. I think this can be done with Domain and Dispatcher module. I have written some of the config to handle registration, but when a call(INVITE) comes in it is not
I offer VOIP telephony services and development of telecommunication solutions, I need a logo and a brochure for my site currently Im using logo which is n...logo and a brochure for my site currently Im using logo which is not mine. Send sample on your propoersal based on the description of my needs This is my site [log ind for at se URL]
We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.
I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a
I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please let me ...
Fix an existing Asterisk PBX system with WebRTC. Right now calls don't come in because FreePBX/Asterisk doesn't register with DID supplier. To make it register, some changes should be made. The system consists of 3 servers, Apache, Asterisk and MySQL. You must be able to work with Teamviewer !
We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [log ind for at se URL] regards
Create and train italian acoustic model based on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should...on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should be configured for telephone (8 khz), in order to integrate it in Asterisk with mrcp server.
...a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows. The server side it's done and use Asterisk server with PJSIP. I would like that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features in the 1st step of the project:
...someone to handle the very few tech support calls I may get while out of town. Might go the whole week with none sometimes. You need to be very knowledgeable of Freepbx and asterisk in order to support my clients. I will on occasion need projects handled such as repartitioning, adding a logical volume etc... Trust is a key because I will be giving you
Hi, I need an asterisk guru to help me acheive my goal. When someone reach my voicemail, I would like my to get a notification on my pager. That mean, system would have to dial a number, wait a couple sec, and dial 123 then hang up. This need to happen everytime someone leave me a voicemail.
Hi, I am looking for Asterisk PBX expert. The person should have through knowledge of how to install configure and maintain Asterisk on linux server. We have 2 projects for which we are looking for an expert. You will be working with me directly to estimate the project and after implementation you will handover the project and explain entire project
This absolutely takes a programmer. After doing all the updates needed for WebRTC and changing OS on Server...OS on Servers, not much is working any more. So the system has to be brought back from near death. This is a system with WebRTC and 3 physically separated servers, Apache, Asterisk, MySQL, plus a commercial ZBF router. Nothing for beginners !
...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud
Hello, I have installed the SuiteCRM and Asterisk system, I need to integrate these systems (call recording, connection identification, history of connections, etc.). How much will it cost a script and configuration?
we did integration between zoho crm with yeastar pbx "u100", and it was working successfully, but if we received any calls through our queue "sales queue or technical queue" it is not displayed calling id name, it displayed pbx trunk number so we would to make a custimze code which enable us to knows who is calling.
I need someone for helping install cluster on vicidial using asterisk and database servers.
basically need to get data from a web service and then read it to caller based on the dialplan. There is an IVR already built where one of the options go to this script and execute the web service. Asterisk 13 CentOS 7 See webservice example attached in a file, so this is a example section of this: <nombreLoteria>DORADO MANANA</nombreLoteria> <
...contemporary logo for or telecommunications and IT conusltancy business. Our business name is Asterswitch, coming from the two main platforms we offer support/consultancy with (i.e. ASTERisk and freeSWITCH) and it is a one man team. Our business provides help with PBX and network solutions. The main objective of the logo: - must be a simple, elegent, professional
...Yuriy M., This is Ferit, I am software developer in Turkey. working in software develepment company provides callcenter solutions in istanbul. I need to learn voip,asterisk from an asterisk guru :) so far i found concept [log ind for at se URL] also i need to learn apis,ami,agi,webrtc etc. Can you plan a learning and practiceing program
Looking for assistance modifying an asterisk script we use for monitoring calls. The current script you dial 556 for Chanspy and it prompts you to enter a specific extension to monitor. This works great. When you are finished monitoring the extension you have to hang up and dial 556 again to monitor a different extension. We would like the script
Hello, I want someone who can install and configure the Asterisk with FreePBX in CentOS Server. I don't want any custom feature for now but all the features which FreePBX and Asterisk offer should be configured properly. Thanks
Features : vTiger Click To Call Integration. vTiger Call Recording. vTiger-ViciDial Lead Synchronization. vTiger-ViciDial User Synchronization. vTiger-ViciDial Group Synchronization. vTiger-ViciDial List Synchronization. vTiger-ViciDial Campaign Synchronization. Call Disposition Synchronization.
I need a script to change the IMEI numbers from asterisk dongles. I'm using Huaway E1553. Can anyone help me in building this? Thanks
To Develop an API for Asterisk to provide a bridge between users and service providers without each of them see the others’ phone number. The receiver of the call should see Asterisk Caller ID we have set (subject to our carrier) or our office number, instead of the caller personal phone number. This is very similar to Masked Phone Numbers service,
I am looking for a Asterisk developer to quickly update existing Asterisk code for dialer, inbound/ outbound/ press 1, and other advanced features. Need to configure advanced features as follows and show how to use. Initial work / payment, and possible ongoing work. Voice Broadcasting Interactive Voice Broadcasting ( Press-1 Campaign) Multi- Campaign
Hi, we need freelancers who are expertise in asterisk or [log ind for at se URL] and have sound knowledge of IVR and other communication features. Candidate should have vicidial expertise too. We will need for 8 Months work, include your monthly price in bid. thank you.
I'm looking for a tech who already has completed a Ringless Voicemail drop system. We will target users in US only so will calli...like Twilio, or FreeSwitch etc..If you Already have completed a Ringless voicemail drop system please bid. Please include the past project info. Thanks Skills required: Asterisk PBX, Telecommunications Engineering, VoIP
...directly on the number you want to transfer a call to directly in our system via the browser. Our agents use either snom phones or Bria softphones. NOTICE that we have an asterisk 11.25.3 server today, where we have setup both agents, queues, extensions etc, which must all be working after this change. There must not be any downtime in connection with
i need someone experienced about issabel (formerly elastix) to set up an maintain our vps based sip switches. serving to multiple clients using multiple sip operators,trunk or sip user based.
...CAPITALS. 3. If you have any notes of your own, please write them at the end of the line, and highlight in blue. /* Important: Please write any notes in between the slash asterisk parenthesis like this one. Failure to do this will ruin the rest of the code. */ 4. Remember that this is an iPhone app, so space is often limited. Please try to keep
To integrate new FreePBX / Asterisk into network with Apache and MySQL Servers. For inbound call centre that uses WebRTC to connect agents over their browser to the system. Must have excellent knowledge of Asterisk with WebRTC.